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Side by Side Diff: webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h

Issue 2086633002: Revert of Remove audio/video distinction for probe packets. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 6 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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51 assert(false); 51 assert(false);
52 } 52 }
53 53
54 // Called for each incoming packet. Updates the incoming payload bitrate 54 // Called for each incoming packet. Updates the incoming payload bitrate
55 // estimate and the over-use detector. If an over-use is detected the 55 // estimate and the over-use detector. If an over-use is detected the
56 // remote bitrate estimate will be updated. Note that |payload_size| is the 56 // remote bitrate estimate will be updated. Note that |payload_size| is the
57 // packet size excluding headers. 57 // packet size excluding headers.
58 // Note that |arrival_time_ms| can be of an arbitrary time base. 58 // Note that |arrival_time_ms| can be of an arbitrary time base.
59 virtual void IncomingPacket(int64_t arrival_time_ms, 59 virtual void IncomingPacket(int64_t arrival_time_ms,
60 size_t payload_size, 60 size_t payload_size,
61 const RTPHeader& header) = 0; 61 const RTPHeader& header,
62 bool was_paced) = 0;
62 63
63 // Removes all data for |ssrc|. 64 // Removes all data for |ssrc|.
64 virtual void RemoveStream(uint32_t ssrc) = 0; 65 virtual void RemoveStream(uint32_t ssrc) = 0;
65 66
66 // Returns true if a valid estimate exists and sets |bitrate_bps| to the 67 // Returns true if a valid estimate exists and sets |bitrate_bps| to the
67 // estimated payload bitrate in bits per second. |ssrcs| is the list of ssrcs 68 // estimated payload bitrate in bits per second. |ssrcs| is the list of ssrcs
68 // currently being received and of which the bitrate estimate is based upon. 69 // currently being received and of which the bitrate estimate is based upon.
69 virtual bool LatestEstimate(std::vector<uint32_t>* ssrcs, 70 virtual bool LatestEstimate(std::vector<uint32_t>* ssrcs,
70 uint32_t* bitrate_bps) const = 0; 71 uint32_t* bitrate_bps) const = 0;
71 72
72 // TODO(holmer): Remove when all implementations have been updated. 73 // TODO(holmer): Remove when all implementations have been updated.
73 virtual bool GetStats(ReceiveBandwidthEstimatorStats* output) const { 74 virtual bool GetStats(ReceiveBandwidthEstimatorStats* output) const {
74 return false; 75 return false;
75 } 76 }
76 77
77 virtual void SetMinBitrate(int min_bitrate_bps) = 0; 78 virtual void SetMinBitrate(int min_bitrate_bps) = 0;
78 79
79 protected: 80 protected:
80 static const int64_t kProcessIntervalMs = 500; 81 static const int64_t kProcessIntervalMs = 500;
81 static const int64_t kStreamTimeOutMs = 2000; 82 static const int64_t kStreamTimeOutMs = 2000;
82 }; 83 };
83 84
84 } // namespace webrtc 85 } // namespace webrtc
85 86
86 #endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMA TOR_H_ 87 #endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMA TOR_H_
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